TWI497966B - Computer-readable memory and device for associating a telephone call with a dialog based on a computer protocol - Google Patents

Computer-readable memory and device for associating a telephone call with a dialog based on a computer protocol Download PDF

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TWI497966B
TWI497966B TW102104434A TW102104434A TWI497966B TW I497966 B TWI497966 B TW I497966B TW 102104434 A TW102104434 A TW 102104434A TW 102104434 A TW102104434 A TW 102104434A TW I497966 B TWI497966 B TW I497966B
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message
call
callee
caller
telephone
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TW201330570A (en
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Amritansh Raghav
Parag Samdadiya
Danny Levin
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Microsoft Technology Licensing Llc
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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L12/00Data switching networks
    • H04L12/66Arrangements for connecting between networks having differing types of switching systems, e.g. gateways
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/42382Text-based messaging services in telephone networks such as PSTN/ISDN, e.g. User-to-User Signalling or Short Message Service for fixed networks
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1101Session protocols
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/10Architectures or entities
    • H04L65/102Gateways
    • H04L65/1023Media gateways
    • H04L65/1026Media gateways at the edge
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/10Architectures or entities
    • H04L65/102Gateways
    • H04L65/1023Media gateways
    • H04L65/103Media gateways in the network
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/10Architectures or entities
    • H04L65/102Gateways
    • H04L65/1033Signalling gateways
    • H04L65/1036Signalling gateways at the edge
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/10Architectures or entities
    • H04L65/102Gateways
    • H04L65/1033Signalling gateways
    • H04L65/104Signalling gateways in the network
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1069Session establishment or de-establishment
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1101Session protocols
    • H04L65/1104Session initiation protocol [SIP]
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M7/00Arrangements for interconnection between switching centres
    • H04M7/0024Services and arrangements where telephone services are combined with data services
    • H04M7/0027Collaboration services where a computer is used for data transfer and the telephone is used for telephonic communication
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M2203/00Aspects of automatic or semi-automatic exchanges
    • H04M2203/65Aspects of automatic or semi-automatic exchanges related to applications where calls are combined with other types of communication
    • H04M2203/652Call initiation triggered by text message

Description

用於根據一電腦協定將一電話通話與一對話相關聯之電腦可讀取記憶體及裝置Computer readable memory and device for associating a telephone call with a conversation in accordance with a computer agreement 對相關申請案之交叉參照Cross-reference to relevant applications

本專利申請案係為2005年8月15日提交的、名稱為「ASSOCIATING A TELEPHONE CALL WITH A DIALOG BASED ON A COMPUTER PROTOCOL SUCH AS SIP」之美國專利申請案No.11/205,762的接續案,該母案之美國專利號為No.8,155,294,公告日期2012年4月10日。該專利申請母案在此以引用方式將其整體併入本文。This patent application is a continuation of U.S. Patent Application Serial No. 11/205,762, filed on Aug. 15, 2005, entitled "ASSOCIATING A TELEPHONE CALL WITH A DIALOG BASED ON A COMPUTER PROTOCOL SUCH AS SIP. The US patent number of the case is No. 8,155,294, and the announcement date is April 10, 2012. This patent application is incorporated herein by reference in its entirety.

本發明係有關於用於根據一電腦協定(像是SIP)將一電話通話與一對話相關聯的方法及電腦可讀取媒體。The present invention relates to a method and computer readable medium for associating a telephone call with a conversation in accordance with a computer protocol (such as SIP).

在公眾交換電話網路(PSTN)中通常是可利用發訊以設定各電路、終結各對話等等。例如,用於PSTN之各種協定的其中一者稱為「共用頻道發訊系統7(SS7)」。在PSTN中,現有的發訊協定可提供對於基本通話設定及卸除、帳務、無線服務及漫遊、本地號碼可攜性、免付費通話服務、發話者ID、三方通話、通話轉接強化特徵等等的支援。然而,這些現 有的發訊協定僅提供有限的支援。例如,對於發話者ID來說,並不保證發話者ID資訊將能夠端對端地行旅。另一個問題是在於若該通話發自於一私有分機交換器(PBX)延伸等等,則該發話者ID對於一使用者而言並不具有唯一性。例如,當一Acme公司雇員發出一通電話時,或會將這通電話識別成「Acme」而非該雇員。In the Public Switched Telephone Network (PSTN), it is common to use messaging to set up circuits, terminate conversations, and the like. For example, one of various protocols for PSTN is referred to as "Shared Channel Messaging System 7 (SS7)." In the PSTN, existing messaging protocols provide basic call setup and disassembly, accounting, wireless services and roaming, local number portability, pay-free calling services, caller ID, three-way calling, and call forwarding enhancement features. Waiting for support. However, these are now Some communication agreements only provide limited support. For example, for the Talker ID, there is no guarantee that the Talker ID information will be able to travel end-to-end. Another problem is that if the call originates from a private extension switch (PBX) extension, etc., the talker ID is not unique to a user. For example, when an Acme employee makes a call, the call may be identified as "Acme" instead of the employee.

而不同於PSTN發訊協定,「對話啟動協定(SIP)」並不受限於在設定發訊階段之過程中的通訊。SIP是一種應用層控制協定,電腦系統可利用來逐一探索並且為以建立、修改與終結多媒體對話。例如,SIP為用以實作出「語音IP」技術之關鍵性協定的其中一者。實作「語音IP」技術通常會牽涉到將語音資訊轉換成數位形式,並且在各離散封包內,而非按一傳統的PSTN電路相涉之協定,予以發送。SIP亦與實作出像是即時傳訊與其他「即時性」通訊技術之各項特徵相關連。例如,一即時傳訊服務可讓參與者發送各項訊息,並且由在一會談中的其他參與者在一兩秒內接收到這些訊息。然後,這些收訊參與者可按一類似方式將回應性訊息發送給其他的參與者。Unlike the PSTN protocol, the Session Initiation Protocol (SIP) is not limited to communication during the setup phase. SIP is an application layer control protocol that computer systems can use to explore and build multimedia conversations. For example, SIP is one of the key agreements for making "voice over IP" technology. Implementing "voice over IP" technology typically involves translating voice information into digital form and transmitting it in discrete packets rather than in accordance with a convention associated with a traditional PSTN circuit. SIP is also associated with real-life messaging and other "immediate" communication technologies. For example, an instant messaging service allows participants to send messages and receive them in a second or two by other participants in a conversation. These recipients can then send responsive messages to other participants in a similar manner.

SIP是一種網際網路提議標準。其規格文件,「RFC 3261」,可自http://www.ietf.org/rfc/rfc3261.txt上取得。其中一份有關於事件通知的SIP延伸規格 文件,「RFC 3265」,可自http://www.ietf.org/rfc/rfc3265.txt處取得。一SIP網路包含可按如一客戶端、伺服器或兩者參與對話的各項實體。SIP支援達四種型式的實體:使用者代理器、代理伺服器、重導伺服器及登錄器。使用者代理器可藉由與其他SIP實體交換訊息以啟動並終結對話。一使用者代理器可為一使用者代理器客戶端(UAC),此者為一啟動SIP請求的裝置,或是一使用者代理器伺服器(UAS),此者為一接收SIP請求並回應於該等請求的裝置。即如範例,「IP電話」、個人數位助理以及任何其他型式的計算裝置皆可作為使用者代理器。某一裝置在一對話裡可為一UAC,而在另一對話內為一UAS,或是可在對話過程中改變角色。一代理伺服器係一可對各客戶端作為一伺服器,並且對於各伺服器為一客戶端的實體。如此進行,代理伺服器可於各UAC與各UAS之間截收、解譯或轉傳訊息。一重導伺服器可接受一SIP請求,並且產生一可將先前發送該請求之UAC導引接觸一替代性網路資源的回應。一登錄器係一伺服器,此者可接受來自於各使用者代理器的登註資訊,並且通知所接收之登註資訊的位置服務。SIP is an Internet proposal standard. Its specification document, "RFC 3261", is available at http://www.ietf.org/rfc/rfc3261.txt. One of them has a SIP extension specification for event notification The document, "RFC 3265", is available at http://www.ietf.org/rfc/rfc3265.txt. A SIP network contains entities that can participate in a conversation as a client, server, or both. SIP supports up to four types of entities: user agent, proxy server, redirection server, and logger. The user agent can initiate and terminate the conversation by exchanging messages with other SIP entities. A user agent can be a User Agent Client (UAC), which is a device that initiates a SIP request, or a User Agent Server (UAS), which receives and responds to a SIP request. For such requested devices. As an example, an "IP phone", a personal digital assistant, and any other type of computing device can be used as a user agent. A device can be a UAC in one conversation and a UAS in another conversation, or can change roles during a conversation. A proxy server is a client for each client and a client for each server. In this way, the proxy server can intercept, interpret or transfer messages between each UAC and each UAS. A redirect server can accept a SIP request and generate a response that can direct the UAC that previously sent the request to an alternate network resource. A logger is a server that accepts the posting information from each user agent and notifies the location service of the received posting information.

SIP支援兩種訊息型式:請求,這些是自一UAC發送至一UAS;以及回應,這些則是在當回應於一請求時從一UAS發送至一UAC。而一SIP訊息是由 三個部分所組成。一SIP訊息的第一部分為「請求列」,其中含有為以表示一訊息方法(即如INVITE)的各欄位,以及一「Request URI」,此者可識別出將該請求所導引至之使用者或服務。該SIP訊息的第二部分含有各標頭,其數值係按如各名稱-數值組對所代表。該SIP訊息的第三部分為該訊息的本體,這是用以描述該待予啟動之對話,或是含有與該對話相關聯的資料。訊息本體可出現在各請求或回應內。SIP supports two types of messages: requests, which are sent from a UAC to a UAS; and responses, which are sent from a UAS to a UAC when responding to a request. And a SIP message is made up of It consists of three parts. The first part of a SIP message is a "request column" containing fields for representing a message method (ie, INVITE) and a "Request URI" that identifies the request to which the request is directed. User or service. The second part of the SIP message contains headers whose values are represented by pairs of name-value pairs. The third part of the SIP message is the body of the message, which is used to describe the conversation to be initiated or contains information associated with the conversation. The message body can appear in each request or response.

茲提供一種為以將一電話通話關聯於一電腦對話,像是對話啟動協定(SIP),的方法及系統。該對話可供以在一計算裝置處提供關聯於一電話通話的資訊。A方想要發話給B方。A及B雙方皆擁有計算裝置及電話。利用彼等計算裝置之間的傳訊處理,A方發送一請求以開始一與B方的電話通話。若該請求為B方的計算裝置所接受,則可建立一電腦對話的發訊階段(即如一SIP邀請交易對話),並且嘗試透過一公眾交換電話網路(PSTN)啟動一電話通話。當收到該電話通話成功建立的確認訊息之後,假定確經適當組態設定,A方的計算裝置及B方的計算裝置即開始一通話階段電腦對話(即如一SIP對話)。可利用該電腦對話(在發訊階段及通話階段兩者 之下)來發送有關於A方與B方識別資料、電話通話之狀態、關聯於該電話通話之各裝置的狀態等等訊息。A method and system for associating a telephone call with a computer conversation, such as a Session Initiation Protocol (SIP), is provided. The dialog can be used to provide information associated with a telephone call at a computing device. Party A wants to send a message to Party B. Both A and B have computing devices and telephones. Using the communication processing between their computing devices, party A sends a request to initiate a telephone conversation with party B. If the request is accepted by the B-party computing device, a messaging phase of the computer conversation (i.e., a SIP Inviting Transaction Dialogue) can be established and an attempt to initiate a telephone conversation over a Public Switched Telephone Network (PSTN). After receiving the confirmation message that the telephone call is successfully established, assuming that the configuration is properly configured, the computing device of the A party and the computing device of the B party start a conversation session (ie, a SIP conversation). Can use this computer conversation (both during the messaging phase and during the conversation phase) Bottom) to send information about the A party and B party identification data, the status of the phone call, the status of each device associated with the phone call, and the like.

有許多系統元件可供以設定對話,這些可,或無需,包含在A方或B方計算裝置上的通訊應用程式;作用中的通訊伺服器,此者可輔助於A方與B方計算裝置之間傳通訊息;於公眾交換電話網路與該等計算裝置之間的各式介面;該公眾交換電話網路(這可包含PBX)的各元件等等。There are a number of system components available for setting up conversations, which may or may not be included in the communication application on the A or B party computing device; the active communication server, which may assist the A and B party computing devices Communication between the public; the various interfaces between the public switched telephone network and the computing devices; the components of the public switched telephone network (which may include PBXs), and the like.

本「概述」係經提供以介紹一種按一較簡化形式,而在後文「詳細說明」所進一步描述的選擇概念。本「概述」並非為以識別所主張之主題項目的各項關鍵特點或基本特徵,亦非為以用於決定所主張主題項目之範圍的輔助。This "Overview" is provided to introduce a selection concept that is further described in a more simplified form and described in the "Detailed Description" below. This “Overview” is not intended to identify key features or essential features of the claimed subject matter, nor is it used to assist in determining the scope of the claimed subject matter.

茲提供一種為以將一電話通話關聯於一電腦式對話之設施(即如軟體設施及相關系統)。在一些具體實施例裡,可併同於一在「一般舊式電話網路」,像是公眾交換電話網路(PSTN),上所撥發之電話通話,而建立起一端對端(end-to-end)SIP對話。然後,該設施可將該PSTN通話狀態同步於SIP對話狀態。例如,該發話方可利用該SIP對話來向該受話方進行識別。這可讓受話方能夠識別出該發話方,且即 使是在當該通話並未含有ID資訊時亦然。此外,可利用一或更多欄位來延伸SIP,藉以支援PSTN狀態對SIP狀態的同步化作業。此等延伸的範例包含一或更多的通話資訊欄位(即如透過一SIP INVITE所發送)、一或更多的通話資訊回應欄位(即如透過SIP 200 OK所發送),以及一或更多的通話資訊相符欄位(即如透過SIP ACK所發送)。A facility (i.e., a software facility and related system) for associating a telephone conversation with a computerized conversation is provided. In some embodiments, the end-to-end (end-to-) may be established in conjunction with a telephone call placed on a "general old telephone network", such as the Public Switched Telephone Network (PSTN). End) SIP conversation. The facility can then synchronize the PSTN call state to the SIP dialog state. For example, the Talker can utilize the SIP dialog to identify the recipient. This allows the recipient to recognize the utterer, and This is also true when the call does not contain ID information. In addition, one or more fields can be used to extend the SIP to support the synchronization of the PSTN state to the SIP state. Examples of such extensions include one or more call information fields (ie, sent via a SIP INVITE), one or more call information response fields (ie, sent via a SIP 200 OK), and one or More call information matches the field (ie, sent via SIP ACK).

在一些具體實施例裡,當一使用者(發話者)想要發出一通話給某位經名列在其電腦聯絡人列表內之人士(受話方)時,此人可選擇該聯絡人,並且表明想要撥出一通電話給這位聯絡人。此時,一在該發話方之計算裝置上的應用程式可將一請求(即如一SIPINVITE)發送至該受話方的計算裝置,而開始一發訊階段對話。該請求可包含該電話資訊係一建立起與該聯絡人之對話的特徵之表示。該受話方此時知悉而預期一通來自於該發話方的電話通話。若該受話方的計算裝置回覆以接受,並表明此者可支援該電話資訊,則該發話方的計算裝置可透過電話系統發出一通話,然後等待直到此者獲得一表示在該受話方電話處開始鈴響的回遞訊息。一旦受話方聽到電話鈴響之後,該受話方可預期該通話將會是來自於該發話方,而這會是成立的,除非是就在接收發出該SIP訊息之發話方的通話前恰獲接一通來自另一發話方的電話。當發話者的計算裝置接收到一該通 話獲得應答並且既已進行一電腦交換表示後,該發話者的計算裝置知曉該PSTN電話通話已然成功,同時發送一表示成功的電腦訊息。該受話方在此刻會接著知曉此一通話是來自於所預期的發話者。In some embodiments, when a user (talker) wants to place a call to a person (receiver) who is listed in his computer contact list, the person can select the contact, and Indicates that you want to make a call to the contact. At this point, an application on the Talker's computing device can send a request (ie, a SIP INVITE) to the recipient's computing device to initiate a messaging phase session. The request may include the telephone information being an indication of the characteristics of the conversation with the contact. The callee is aware of this and expects a telephone call from the Talker. If the receiving party's computing device replies with acceptance and indicates that the person can support the telephone information, the Talker's computing device can place a call through the telephone system and then wait until the person obtains a representation at the called party's telephone A reply message that starts ringing. Once the callee hears the phone ringing, the caller can expect that the call will come from the caller, and this will be true unless it is just before the call from the caller who sent the SIP message. A call from another caller. When the talker's computing device receives a pass After the call is answered and a computer exchange representation has been made, the talker's computing device knows that the PSTN telephone call has succeeded and simultaneously sends a computer message indicating success. The callee will then know at this point that the call is from the intended talker.

在該通話獲得應答後即收到一確認訊息,而建立起一通話階段電腦對話。從而,可在發話者計算裝置及受話方計算裝置之間往返發送資訊。此項資訊可包含該通話已經被保留(hold)、偏轉(deflected)、傳輸(transfered)的表示。After the call is answered, a confirmation message is received, and a conversation phase computer conversation is established. Thus, information can be sent back and forth between the talker computing device and the callee computing device. This information may include a representation that the call has been held, deflected, or transferred.

第1圖為一方塊圖,其中說明在一具體實施例中該設施的各項元件。在一些具體實施例裡,該設施含有一作用中通訊伺服器100,此者可透過網際網路135而連接於一或更多的客戶端150及175。該作用中通訊伺服器100含有一存現(presence)伺服器105,此者可為一維護有關於即時傳訊服務之各使用者的狀態資訊之SIP登錄器(registrar)。該存現伺服器105可含有一訂閱(subscription)元件115及一存現文件存庫120。該訂閱元件115可對於各使用者維護一經訂閱於該使用者存現資訊的目前使用者列表。對於各使用者的狀態資訊可為按一稱為存現文件之文件的形式所表現。該存現文件存庫120含有對於各使用者,而表示出該使用者在該作用中通訊伺服器100上之目前狀態的存現文件。Figure 1 is a block diagram showing the various components of the facility in a particular embodiment. In some embodiments, the facility includes an active communication server 100 that is coupled to one or more clients 150 and 175 via Internet 135. The active communication server 100 includes a presence server 105, which can be a SIP registrar that maintains status information about each user of the instant messaging service. The deposit server 105 can include a subscription component 115 and a deposit file repository 120. The subscription component 115 can maintain, for each user, a current user list that is subscribed to the user's presence information. The status information for each user can be expressed in the form of a file called a deposit file. The stored file repository 120 contains a stored file representing the current state of the user on the communication server 100 for the user.

一使用者資料伺服器110含有對於各使用者的 永續性(persistent)資料,並且含有一聯絡人列表存庫125。該聯絡人列表存庫125含有一各使用者之聯絡人的列表。在一些具體實施例裡,該作用中通訊伺服器100可將一使用者之聯絡人列表提供給該使用者的經登錄端點。一此等端點的範例包含客戶端A 150及客戶端B 175,此二者可經關聯於各人類使用者。在一些具體實施例裡,該等客戶端150及175含有通訊應用程式155。該通訊應用程式155可包含一訂閱元件160、一登錄元件165以及一使用者介面元件170。該訂閱元件160可訂閱該使用者之聯絡人的存現資訊。該登錄元件165可將該使用者的端點登錄於該作用中通訊伺服器100,並且將各更新項目發布至該使用者的存現資訊。該使用者介面元件170可在當與該通訊應用程式155進行互動時,提供對該使用者所呈現的視窗、檢視及其他圖形要件。A user profile server 110 contains for each user Persistent data and contains a contact list repository 125. The contact list repository 125 contains a list of contacts for each user. In some embodiments, the active communication server 100 can provide a user's contact list to the user's logged in endpoint. An example of such an endpoint includes client A 150 and client B 175, which may be associated with each human user. In some embodiments, the clients 150 and 175 include a communication application 155. The communication application 155 can include a subscription component 160, a login component 165, and a user interface component 170. The subscription component 160 can subscribe to the presence information of the contact of the user. The login component 165 can log the user's endpoint to the active communication server 100 and post each update item to the user's presence information. The user interface component 170 can provide windows, views, and other graphical requirements presented to the user when interacting with the communication application 155.

該設施可經關聯於一PSTN網路130,而與各客戶端相關聯的電話(180及190)則可經此網路以進行通訊。該等電話180及190各者亦可經關聯於一傳統的PBX 185及195。一遠端通話控制(RCC)伺服器196可,依需要而定,提供一於該PBX(或其他PSTN元件)與該客戶端計算裝置之間的介面。按此方式,即可令該客戶端計算裝置知悉在其中發出一通話之PBX處所進行的活動。例如,該遠端通話控制伺服器可作為一閘道器,藉以將該通訊應用程式所使用 的標準協定轉換成由該PBX所支援的特定電腦電話整合(CTI)協定。雖未圖示,然一或更多的閘道器元件可有助於透過該PSTN 130進行通訊。例如,可提供一SIP-PSTN閘道器以有助於一計算裝置與一電話之間的通話。The facility can be associated with a PSTN network 130, and the phones (180 and 190) associated with each client can communicate over the network. Each of the phones 180 and 190 can also be associated with a conventional PBX 185 and 195. A remote call control (RCC) server 196 can, if desired, provide an interface between the PBX (or other PSTN component) and the client computing device. In this manner, the client computing device can be made aware of the activities performed at the PBX where a call is placed. For example, the remote call control server can act as a gateway for use by the communication application. The standard agreement is translated into a specific computer telephony integration (CTI) agreement supported by the PBX. Although not shown, one or more of the gateway components can facilitate communication through the PSTN 130. For example, a SIP-PSTN gateway can be provided to facilitate a call between a computing device and a telephone.

在該設施上所實作之各計算裝置可包含一中央處理單元、記憶體、輸入裝置(即如鍵盤及點指裝置)、輸出裝置(即如顯示裝置),以及儲存裝置(即如碟機)。該記憶體及儲存裝置為電腦可讀取媒體,此者可含有實作該系統的各項指令。此外,可將該等資料結構與訊息結構加以儲存,或是透過一像是在一通訊鏈結上之信號的資料傳輸媒體而進行傳送。可使用各種通訊鏈結,像是網際網路、區域網路、廣域網路、點對點撥接式連接、行動電話網路等等。Each computing device implemented on the facility may include a central processing unit, a memory, an input device (ie, a keyboard and a pointing device), an output device (ie, a display device), and a storage device (ie, a disk drive) ). The memory and storage device is a computer readable medium, which may contain instructions for implementing the system. In addition, the data structures and message structures can be stored or transmitted through a data transmission medium such as a signal on a communication link. Various communication links can be used, such as internet, regional network, wide area network, point-to-point dial-up connection, mobile phone network, and the like.

可在各種操作環境下實作該設施的具體實施例,而這些包含個人電腦、伺服器電腦、手持式或膝上型裝置、多重處理器系統、微處理器式系統、可程式化消費性電子裝置、數位相機、網路電話通話、迷你電腦、主機電腦、含有上述系統或裝置之任者的分散式計算環境等等。各電腦系統可為行動電話、個人數位助理、智慧型手機、個人電腦、可程式化消費性電子裝置、數位相機等等。Specific embodiments of the facility can be implemented in a variety of operating environments, including personal computers, server computers, handheld or laptop devices, multi-processor systems, microprocessor-based systems, and programmable consumer electronics Devices, digital cameras, VoIP calls, mini computers, host computers, decentralized computing environments containing any of the above systems or devices, and the like. Each computer system can be a mobile phone, a personal digital assistant, a smart phone, a personal computer, a programmable consumer electronic device, a digital camera, and the like.

該系統可按如各電腦可執行指令,像是程式模組,而由一或更多電腦或其他裝置執行之一般情境 所加以描述。一般說來,各程式模組包含可執行特定任務或是實作出特定抽象資料型態之副程式、程式、物件、元件、資料結構等等。通常,可按如各種具體實施例中所需要將各程式模組的功能性予以合併或分散。The system can be executed by one or more computers or other devices in a general context, such as computer-executable instructions, such as program modules. Described. In general, each program module contains subroutines, programs, objects, components, data structures, etc. that can perform specific tasks or implement specific abstract data types. In general, the functionality of the various program modules may be combined or distributed as desired in various embodiments.

第2圖為一網路圖,其中顯示一在兩個客戶端(像是第1圖的客戶端A 150及客戶端175)間,於該SIP對話建立程序200之中的通訊流程(即如各項含有方法及回應的訊息),在此該客戶端A透過PSTN呼叫該客戶端B。在進行第2圖的通訊流程時並未出現競賽情況(像是因嘗試撥發通話之第三方所造成的競賽)。在所述具體實施例裡,是透過CSTA ECMA-269通話控制抽象化(abstraction)(由各計算應用程式所使用以監視並控制像是PBX的交換系統)來進行PSTN發訊(利用虛線箭頭所代表)。在一些具體實施例裡,是在個別的PBX處實作此發訊處理,並且或可關聯於一遠端通話控制伺服器。雖經描繪如從客戶端行旅至客戶端,然第2圖內所敘述的SIP訊息(亦即按實線所顯示者)可另為透過一像是第1圖之作用中通訊伺服器100的共用伺服器所傳通。時間是朝該圖下方進移,使得該圖頂部處的訊息在時間上會比起底部者為較早發送。Figure 2 is a network diagram showing a communication flow between the two clients (such as client A 150 and client 175 of Figure 1) in the SIP dialog establishing program 200 (i.e., Each message contains a method and a response, where the client A calls the client B through the PSTN. There was no competition in the communication process of Figure 2 (such as a competition caused by a third party attempting to make a call). In the specific embodiment, the CST ECMA-269 call control abstraction (used by each computing application to monitor and control a switching system such as a PBX) for PSTN signaling (using a dotted arrow) representative). In some embodiments, the signaling process is implemented at an individual PBX and may be associated with a remote call control server. Although depicted as traveling from the client to the client, the SIP message (that is, displayed by the solid line) described in FIG. 2 can be additionally transmitted through the communication server 100 as shown in FIG. The shared server is transmitted. The time is moved toward the bottom of the graph so that the message at the top of the graph will be sent earlier than the bottom.

該處理程序開始於其中該客戶端A發送一SIP INVITE 205,並且連同一標準SIP Invite標頭及一通 話資訊標頭。在此訊息205中的各欄位可包含一發話者電話通用資源識別碼(URI)、一受話電話URI、一選用發話者ID(像是一公司號碼)、一通話型式(即如普通、計費顧問、會議等)等等。若確有一該客戶端A與該客戶端B間之SIP連接性的潛在機會,則回應於此,該客戶端B發送一含有一「所支援電話」標頭的SIP 183(進行中)回應210。此項目代表一SIP對話之發訊階段的開始處。此時,該客戶端B可預期收到一來自該客戶端A的通話。若該客戶端A無法透過SIP與該客戶端B進行通訊(即如無可用位址、無聯盟關係等等),則該客戶端A不會收到一SIP 183(進行中),而是將收到一個4xx的回應碼(表示一錯誤)。The process begins with the client A sending a SIP INVITE 205 and the same standard SIP Invite header and one pass. Message header. The fields in the message 205 may include a caller's telephone universal resource identifier (URI), a called telephone URI, a selected caller ID (such as a company number), and a call type (ie, ordinary, Fee consultants, meetings, etc.) and so on. In response to this, there is a potential opportunity for SIP connectivity between the client A and the client B. In response, the client B sends a SIP 183 (in progress) response 210 containing a "supported phone" header. . This project represents the beginning of the messaging phase of a SIP conversation. At this point, the client B can expect to receive a call from the client A. If the client A cannot communicate with the client B through the SIP (ie, if there is no available address, no alliance, etc.), the client A will not receive a SIP 183 (in progress), but will Received a 4xx response code (indicating an error).

次五項通訊(215、220、225、230及235)表示對於該PSTN而進行的發訊。若所使用的是一遠端通訊控制伺服器,則可在該遠端通話控制伺服器與個別客戶端之間傳通這些通訊。或另者,可直接地在一PBX及個別客戶端之間傳通。在所述具體實施例裡,這些通訊的第一個為一「Make Call」服務請求信號215。此信號可為關聯於將一連接ID指配予該發話裝置。然後,該客戶端A收到一「Make Call」回應信號220,其後為一「Originated」信號225。該客戶端B所收到的一「Delivered」信號230(接收自該客戶端B的PBX或RCC)是對應於該客戶端B的 電話鈴響。該客戶端A亦收到一表示電話鈴響事件的「Delivered」信號235。The next five communications (215, 220, 225, 230, and 235) indicate the communications made for the PSTN. If a remote communication control server is used, the communication can be communicated between the remote call control server and individual clients. Alternatively, it can be directly transmitted between a PBX and individual clients. In the particular embodiment, the first of these communications is a "Make Call" service request signal 215. This signal may be associated with assigning a connection ID to the handset. The client A then receives a "Make Call" response signal 220 followed by an "Originated" signal 225. A "Delivered" signal 230 received by the client B (received from the client B's PBX or RCC) corresponds to the client B. The phone rings. The client A also receives a "Delivered" signal 235 indicating a ringing event.

在該客戶端A及該客戶端B兩者既已收到「Delivery」信號之後,該客戶端B發送一SIP 180(鈴響)回應240。此回應可包含一所支援電話標頭。類似地,在一些具體實施例裡,在該客戶端B回應於該客戶端A而發送一SIP 180(鈴響)之前,該客戶端A可先發送一SIP 180(鈴響)回應給該客戶端B,其中含有能夠讓該客戶端B知道是由誰發話的識別資訊。After both client A and client B have received the "Delivery" signal, client B sends a SIP 180 (ringing) response 240. This response can include a support phone header. Similarly, in some embodiments, before client B sends a SIP 180 (ringing) in response to the client A, the client A may first send a SIP 180 (ringing) response to the client. End B, which contains identification information that enables the client B to know who is speaking.

當關聯於該客戶端B的使用者真正地應答該鈴響電話時,該客戶端B發送一SIP 200 OK回應245。除一所支援電話標頭以外,在此回應245內的各欄位亦可包含一發話者電話通用資源識別碼(URI)、一受話電話URI及一狀態欄位。When the user associated with the client B actually answers the ringing phone, the client B sends a SIP 200 OK response 245. In addition to a support phone header, the fields in response 245 may also include a caller's telephone universal resource identifier (URI), a called telephone URI, and a status field.

藉由應答該鈴響電話,關聯於該客戶端B的使用者即啟動進一步的PSTN發訊。例如,該客戶端B及該客戶端A兩者可自其個別的PBX/RCC處,接收到一「Established」信號(分別地為250及255)。由於此一發訊的結果,該客戶端A發送一SIP ACK 260。該SIP ACK 260表示該INVITE交易的結束,以及該發訊階段對話的結束。除標頭資訊以外,此ACK亦可包含有關於一相符旗標的資訊。這可讓該客戶端B能夠將該通話關聯於一SIP URI與一會談 視窗,藉此建立起一SIP通話階段對話。By answering the ringing call, the user associated with the client B initiates a further PSTN call. For example, both Client B and Client A can receive an "Established" signal (250 and 255, respectively) from their respective PBX/RCC. Due to the result of this transmission, the client A sends a SIP ACK 260. The SIP ACK 260 indicates the end of the INVITE transaction and the end of the conversation phase session. In addition to the header information, this ACK can also contain information about a matching flag. This allows the client B to associate the call with a SIP URI and talk to it. Windows, thereby establishing a SIP call phase dialogue.

第3圖為一網路圖,其中顯示一在兩個客戶端(像是第1圖的客戶端A 150及客戶端175)間,於該SIP對話建立程序300之中的通訊流程(即如各項含有方法及回應的訊息),在此該客戶端A透過PSTN呼叫該客戶端B。第3圖的通訊流程是在當出現競賽情況時所進行。更詳細地說,一第三發話者C在該客戶端A嘗試撥話給該客戶端B之相同時間點處,正嘗試撥話給關聯於該客戶端B的電話。即如參照第2圖,在第3圖中的所述具體實施例裡,是透過ECMA-269通話控制抽象化以進行各PSTN發訊(利用虛線箭頭所表示)。在一些具體實施例裡,是在個別的PBX處實作此發訊處理,並且或可關聯於一遠端通話控制伺服器。雖經描述為自客戶端行旅至客戶端,然第3圖內的SIP訊息(即如該等按如實線所表示者)可令為透過一像是第1圖內的作用中通訊伺服器之共用伺服器所傳通。Figure 3 is a network diagram showing a communication flow between the two clients (such as client A 150 and client 175 of Figure 1) in the SIP dialog establishing program 300 (i.e., Each message contains a method and a response, where the client A calls the client B through the PSTN. The communication process in Figure 3 is performed when a competition situation occurs. In more detail, a third caller C is attempting to dial a call associated with the client B at the same point in time when the client A attempts to dial the call to the client B. That is, as shown in Fig. 2, in the specific embodiment of Fig. 3, the ECMA-269 call control abstraction is performed for each PSTN transmission (indicated by a dotted arrow). In some embodiments, the signaling process is implemented at an individual PBX and may be associated with a remote call control server. Although described as traveling from the client to the client, the SIP message in Figure 3 (that is, as indicated by the solid line) can be passed through an active communication server as shown in Figure 1. The shared server is transmitted.

該處理程序開始於其中該客戶端A發送一SIP INVITE 305,並且連同一標準SIP Invite標頭及一通話資訊標頭,如此啟動一SIP發訊對話。回應於此,該B發送一含有一「所支援電話」標頭的SIP 183(進行中)回應310。在次三項通訊裡(315、320及325),該客戶端A在嘗試設定一PSTN通話過程中會與該PBX/RCC進行發訊通訊。然而,在進行之前,該發 話者C既已發話給該客戶端B的電話。從而,當正常地進行該等「Make Call」服務請求信號315、該「Make Call」回應信號320及該「Originated」信號325時,在該C的通話經遞交至該客戶端B電話之後,該客戶端A很快地就會收到一「Failed」信號335。此刻,該客戶端B預期該鈴響電話是來自於該客戶端A。然後該客戶端B可應答該電話,而此會具有一些該發話者並非是該客戶端A的細微風險。因應於此,該客戶端A可發送一SIP CANCEL 340給該客戶端B。此時,該客戶端B獲悉該通話並非是來自於該客戶端A。這會結束建立起該SIP通話階段對話的程序。與此同時,關聯於該客戶端B的使用者應答來自於該發話者C的電話,並且會從該PBX/RCC處收到一對於該C之通話的「Established」信號345。The process begins with the client A sending a SIP INVITE 305 with the same standard SIP Invite header and a call message header, thus initiating a SIP messaging session. In response to this, the B sends a SIP 183 (in progress) response 310 containing a "Supported Phone" header. In the next three communications (315, 320, and 325), the client A will communicate with the PBX/RCC during an attempt to set up a PSTN call. However, before proceeding, the hair The caller C has already sent a call to the client B. Therefore, when the "Make Call" service request signal 315, the "Make Call" response signal 320, and the "Originated" signal 325 are normally performed, after the call of the C is delivered to the client B, the Client A will quickly receive a "Failed" signal 335. At this point, the client B expects the ringing phone to be from the client A. The client B can then answer the call, and there will be some subtle risks that the caller is not the client A. In response to this, the client A can send a SIP CANCEL 340 to the client B. At this point, the client B learns that the call is not from the client A. This will end the process of establishing the SIP talk phase session. At the same time, the user associated with the client B answers the call from the caller C and receives an "Established" signal 345 for the call to the C from the PBX/RCC.

第4圖顯示一在一具體實施例中,一透過該作用中通訊伺服器自一第一SIP客戶端(發話方)發送至一第二SIP客戶端(受話方)之INVITE訊息400的內容。該INVITE訊息400包含一IP標頭405、一TCP標頭410以及一SIP訊息412。對於底層傳輸而言,可使用除TCP/IP以外的協定。該SIP訊息412包含一具SUBSCRIBE方法之請求列415、一「From」標頭420、一「To」標頭425、一「Caller Tel URI」標頭430、一「Called Tel URI」標頭435、一 「Supplemental Caller ID」標頭440、一「Call Type」標頭445、一「Supported」標頭450以及一「Content-Length」標頭455。該「From」標頭420可識別出該發送客戶端計算裝置(該發話者)。該「To」標頭425標定出此一封包的發送者現正訂閱於哪一端點(受話方)。該「Caller Tel URI」標頭430標定出該發話者的電話,並且該「Called Tel URI」標頭435標定出該受話方的電話。該「Supplemental Caller ID」標頭440提供一替代性「Caller ID」(即如與該發話者相關聯之公司的ID),且為選用者。該「Call Type」標頭445標定出該通話的型式,這可例如為一普通通話、一會議通話、一計費顧問通話等等。該「Supported」標頭450標定該發送者所支援的任何特殊行為。該「Content-Length」標頭455具有一數值0,這表示並無資料後隨於該SIP訊息標頭。然而,在一替代性實作裡,有些包含在各標頭(即如「Caller Telephone URI」、「Supplemental Caller ID」)內的資料可另為在該訊息資料之內所提供。Figure 4 shows, in a specific embodiment, the contents of an INVITE message 400 transmitted from a first SIP client (sender) to a second SIP client (recipient) via the active communication server. The INVITE message 400 includes an IP header 405, a TCP header 410, and a SIP message 412. For the underlying transport, protocols other than TCP/IP can be used. The SIP message 412 includes a SUBSCRIBE method request column 415, a "From" header 420, a "To" header 425, a "Caller Tel URI" header 430, and a "Called Tel URI" header 435. One A "Supplemental Caller ID" header 440, a "Call Type" header 445, a "Supported" header 450, and a "Content-Length" header 455. The "From" header 420 identifies the transmitting client computing device (the talker). The "To" header 425 calibrates which endpoint (the recipient) the sender of the packet is currently subscribed to. The "Caller Tel URI" header 430 calibrates the caller's phone, and the "Called Tel URI" header 435 identifies the callee's phone. The "Supplemental Caller ID" header 440 provides an alternate "Caller ID" (i.e., the ID of the company associated with the talker) and is an option. The "Call Type" header 445 calibrates the type of the call, which can be, for example, a normal call, a conference call, a billing advisor call, and the like. The "Supported" header 450 calibrates any special behavior supported by the sender. The "Content-Length" header 455 has a value of 0, which means that there is no data followed by the SIP message header. However, in an alternative implementation, some of the information contained in each header (ie, "Caller Telephone URI", "Supplemental Caller ID") may be provided separately within the message material.

第5圖顯示在一具體實施例裡,一由一受話方(透過該作用中通訊伺服器)發送至一SIP客戶端之INVITE回應500的內容。該INVITE訊息500包含一IP標頭505、一TCP標頭510以及一SIP訊息515。對於底層傳輸而言,可使用除TCP/IP以外的協定。該SIP訊息515包含一SIP回應識別碼520、一「To」 標頭525、一「From」標頭530、一「Caller Tel URI」標頭531、一「Called Tel URI」標頭532、一「Status」標頭533、一「CSeq」標頭535、一「Supported」標頭540以及一「Content-Length」標頭545。該SIP回應識別碼520可識別出該SIP版本2.0,以及表示出該訂閱作業為成功的請求狀態(即如202「Accepted」)。該「To」標頭525標定出(specifies)該請求之原始發送者已經對其而訂閱的端點。該「From」標頭530標定出該請求的原始發送者。該「Caller Tel URI」標頭531標定出該發話者的電話,並且該「Called Tel URI」標頭532標定出該受話方的電話。該「Status」標頭533標定出該受話方之通話系統的狀態。該「CSeq」標頭535經納入於來自該作用中通訊伺服器之每個請求及回覆兩者的訊息內,並且用以作為一決定來自於該伺服器之訊息的相對順序之序號。可對各個SIP對話產生不同的序號,並且該序號通常可標定產生該對話之請求的型式。例如,該「CSeq」標頭535具有一數值1 INVITE,表示此為來自在該等「To」及「From」標頭所標定之各端點間,由一INVITE請求所產生之對話裡該伺服器的第一個訊息。具有一較低序號的訊息是比具有一較高序號者較早自該伺服器所發送。該「Supported」標頭540標定出該發送者所支援的任何特殊行為。該「Content-Length」標頭545具有一 數值0,這表示並後隨於該SIP訊息標頭有0位元的資料。然而,在一替代性實作裡,有些包含在各標頭(即如「Caller Telephone URI」、「Called telephone URL」相符旗標等等)內的資料可另為在該訊息資料之內所提供。Figure 5 shows the contents of an INVITE response 500 sent by a callee (via the active communication server) to a SIP client in one embodiment. The INVITE message 500 includes an IP header 505, a TCP header 510, and a SIP message 515. For the underlying transport, protocols other than TCP/IP can be used. The SIP message 515 includes a SIP response identifier 520, a "To" Header 525, a "From" header 530, a "Caller Tel URI" header 531, a "Called Tel URI" header 532, a "Status" header 533, a "CSeq" header 535, and a "" The Supported header 540 and a "Content-Length" header 545. The SIP response identifier 520 can identify the SIP version 2.0 and indicate the request status of the subscription job as successful (ie, 202 "Accepted"). The "To" header 525 specifies the endpoint to which the original sender of the request has subscribed. The "From" header 530 calibrates the original sender of the request. The "Caller Tel URI" header 531 calibrates the caller's phone, and the "Called Tel URI" header 532 calibrates the callee's call. The "Status" header 533 calibrates the status of the caller's call system. The "CSeq" header 535 is included in the message from each request and reply of the active communication server and is used as a sequence number for determining the relative order of messages from the server. Different serial numbers can be generated for each SIP dialog, and the sequence number can typically calibrate the type of request that generated the dialog. For example, the "CSeq" header 535 has a value of 1 INVITE, indicating that the servo is from the session generated by an INVITE request from among the endpoints calibrated by the "To" and "From" headers. The first message of the device. A message with a lower sequence number is sent from the server earlier than the one with a higher sequence number. The "Supported" header 540 calibrates any special behavior supported by the sender. The "Content-Length" header 545 has a The value 0, which is followed by the 0-bit data of the SIP message header. However, in an alternative implementation, some of the information contained in each header (ie, "Caller Telephone URI", "Called Telephone URL", etc.) may be provided separately within the message material. .

第6圖為一流程圖,其中說明一在該發話方計算裝置處,為以建立起一關聯於一電話通話之SIP對話的例式(routine)600。當一發話方在其客戶端計算裝置處利用一通訊應用程式嘗試撥出一通話時,即啟用該例式600。在方塊605處,該例式600產生並發送一SIP INVITE訊息,這可為例如發送至一作用中通訊伺服器,藉以建立起一與一受話方之客戶端計算裝置的連接。此情境係假定該受話方的客戶端計算裝置經組態設定以進行SIP通訊。若否,則該通話的電話通話特點仍可繼續,但則並未建立起該SIP對話。Figure 6 is a flow diagram illustrating a routine 600 at the Talker Computing Device to establish a SIP dialog associated with a telephone call. The example 600 is enabled when a caller attempts to place a call using a communication application at its client computing device. At block 605, the example 600 generates and transmits a SIP INVITE message, which may be, for example, sent to an active communication server to establish a connection with a client computing device of the recipient. This scenario assumes that the called party's client computing device is configured to perform SIP communications. If not, the phone call feature of the call can continue, but the SIP dialog is not established.

在方塊610處,既已建立起該SIP發訊階段對話,並且該例式自該伺服器處收到一「SIP Session Progress」(183)回應。在方塊615處,該發話方客戶端計算裝置發送並接收一或更多的PSTN信號,這可包含一「Make Call」信號、一「Make Call」回應信號、一「Originated」回應信號、一「Delivery」回應信號等等。即如前述,這些信號可直接地,或是透過一遠端通話控制(RCC)介面,而在該客戶端與該 PBX之間傳通。在決策方塊620處,若在方塊615處的各信號含有一「Failed」回應信號,則這表示該PSTN通話處理失敗,並且該例式繼續到方塊625,藉以發送一SIP取消訊息,而這將會終結任何對建立一SIP對話的進一步嘗試(或另者,繼續該SIP對話,而可在該受話方之客戶端計算裝置處登錄下嘗試進行一通話的事實)。At block 610, the SIP messaging phase session is established and the instance receives a "SIP Session Progress" (183) response from the server. At block 615, the Talker client computing device transmits and receives one or more PSTN signals, which may include a "Make Call" signal, a "Make Call" response signal, an "Originated" response signal, and a " Delivery" response signal and so on. As mentioned above, these signals can be directly or through a remote call control (RCC) interface at the client and the Pass between PBX. At decision block 620, if the signals at block 615 contain a "Failed" response signal, this indicates that the PSTN call processing failed, and the example continues to block 625 to send a SIP cancel message, which will Any further attempts to establish a SIP session (or otherwise, continuing the SIP session, but the fact that a call can be attempted at the client computing device of the called party) will be terminated.

然而,若在決策方塊620處,該等在方塊615的信號內並未含有一「Failed」回應信號,則該例式600繼續到方塊630,在此會收到一SIP鈴響(180)回應。在由該受話方應答該通話之後,在方塊635該例式600收到一SIP OK(200)回應,這表示一成功交易。雖未顯著加以說明,然若該受話方並未應答,則該例式600可最終地獲得一OK回應等待逾時(而非前進到方塊635),令該例式600結束。However, if at decision block 620, the signals in block 615 do not contain a "Failed" response signal, then the example 600 continues to block 630 where a SIP ringing (180) response is received. . After the call is answered by the callee, at block 635 the routine 600 receives a SIP OK (200) response indicating a successful transaction. Although not explicitly illustrated, if the callee does not answer, the example 600 may eventually obtain an OK response wait timeout (rather than proceeding to block 635), causing the example 600 to end.

在方塊640處,該例式600接收一來自該PBX/RCC的PSTN「Established」信號。回應於此,該例式600發送一表示既已建立起一SIP對話的SIP ACK(方塊645)。接著,該SIP對話可在該通話的過程中,或是之後,繼續進行。這可用來發送各項訊息(即如發話者ID訊息、通話偏向訊息、通話等待訊息等等),而無需仰賴於PSTN發訊。At block 640, the example 600 receives a PSTN "Established" signal from the PBX/RCC. In response thereto, the example 600 sends a SIP ACK indicating that a SIP dialog has been established (block 645). The SIP dialog can then continue during or after the call. This can be used to send messages (ie, caller ID messages, call bias messages, call waiting messages, etc.) without relying on PSTN messaging.

第7圖為一流程圖,其中說明一在該受話方計算裝置處,為以建立起一關聯於一電話通話之SIP 對話的例式700。當一發話方在其客戶端計算裝置處利用一通訊應用程式嘗試撥出一通話時,即啟用該例式700。在方塊705處,在該發話方的客戶端既已嘗試建立起一與一受話方之客戶端計算裝置的連接之後,該例式700收到一SIP INVITE訊息,這可為例如發送自一作用中通訊伺服器。此情境係假定該受話方的客戶端計算裝置經組態設定以進行SIP通訊。若否,則該通話的電話通話特點仍可繼續,然並未建立起該SIP對話。Figure 7 is a flow chart illustrating a SIP at the callee computing device for establishing a call associated with a telephone call Example 700 of the conversation. The example 700 is enabled when a caller attempts to place a call using a communication application at its client computing device. At block 705, after the client of the Talker has attempted to establish a connection with a client computing device of a recipient, the Instance 700 receives a SIP INVITE message, which may be, for example, sent from a role. Medium communication server. This scenario assumes that the called party's client computing device is configured to perform SIP communications. If not, the phone call feature of the call can continue, but the SIP dialog is not established.

在方塊710處,該例式700自該伺服器產生並發送一「SIP Session Progress」(183)回應,這表示開始一SIP發訊階段對話。在方塊715處,該例式700收到一「PSTN Delivered」信號,這表示既已在該PSDN上對該受話方發出一通話。此信號可直接地,或是透過一遠端通話控制(RCC)介面,於該例式700與該PBX之間傳通。若在該決策方塊720處該例式收到一SIP Cancel,則這表示該發話方客戶端既已偵測到一PSTN通話建立失敗,並且將不會進行進一步SIP通訊(令該例式700結束)。然而,若是在決策方塊720處並未收到SIP取消訊息,則該SIP通訊繼續到方塊725,在此該例式700產生並發送一SIP鈴響(180)回應。At block 710, the example 700 generates and sends a "SIP Session Progress" (183) response from the server indicating that a SIP signaling phase session is initiated. At block 715, the example 700 receives a "PSTN Delivered" signal indicating that a call has been placed on the PSDN on the PSDN. This signal can be communicated between the example 700 and the PBX either directly or through a remote call control (RCC) interface. If the instance receives a SIP Cancel at the decision block 720, this indicates that the Talker client has detected a PSTN call setup failure and will not perform further SIP communication (let the example 700 end) ). However, if a SIP cancel message is not received at decision block 720, the SIP communication continues to block 725 where the example 700 generates and sends a SIP ring (180) response.

在該受話方既已應答該通話之後,在方塊730處,該例式700產生並發送一SIP OK(200)回應,這 表示既已出現一成功的SIP交易。雖未顯著加以說明,然若該受話方並未應答,則該例式700可最終地獲得一OK回應等待逾時(而非前進到方塊730),而令以該例式700結束。After the called party has answered the call, at block 730, the example 700 generates and sends a SIP OK (200) response, which Indicates that a successful SIP transaction has occurred. Although not explicitly stated, if the callee does not answer, the example 700 may eventually obtain an OK response wait timeout (rather than proceeding to block 730) and terminate with the example 700.

在方塊735處,該例式700接收一來自於該PBX/RCC的PSTN「Established」信號。假定該發話方客戶端亦收到一類似信號,則該例式700收到一SIP ACK,這表示既已建立起一SIP通話階段對話(方塊740)。接著,所建立起的SIP對話可在該通話的過程中,或是之後,繼續進行。這可用來發送各項訊息(即如發話者ID訊息、通話偏向訊息、通話等待訊息等等),而無需仰賴於PSTN發訊。At block 735, the example 700 receives a PSTN "Established" signal from the PBX/RCC. Assuming that the originating client also receives a similar signal, the example 700 receives a SIP ACK indicating that a SIP talk phase session has been established (block 740). The established SIP dialog can then continue during or after the call. This can be used to send messages (ie, caller ID messages, call bias messages, call waiting messages, etc.) without relying on PSTN messaging.

雖既已按照特定語言以結構特徵及/或方法動作來描述該標的,然應了解在後載申請專利範圍中所定義主題事項並非必然地受限於前述各項特定特徵或動作。相反地,上述各項特定特徵及動作係按如實作該申請專利範圍之範例形式所揭示。從而,除如後載之申請專利範圍者外,本發明並無限制。The subject matter is described in terms of structural features and/or methodological acts in a particular language, and it is understood that the subject matter defined in the scope of the appended claims is not necessarily limited to the specific features or acts described. Rather, the specific features and acts described above are disclosed as examples of the scope of the application. Accordingly, the invention is not limited except as claimed in the appended claims.

100‧‧‧作用中通訊伺服器100‧‧‧Active communication server

135‧‧‧網際網路135‧‧‧ Internet

105‧‧‧存現伺服器105‧‧‧Current server

150‧‧‧客戶端A150‧‧‧Client A

110‧‧‧使用者資料伺服器110‧‧‧User data server

175‧‧‧客戶端B175‧‧‧Client B

115‧‧‧訂閱元件115‧‧‧Subscription component

155‧‧‧通訊應用程式155‧‧‧Communication application

120‧‧‧存現文件存庫120‧‧‧Current file repository

160‧‧‧訂閱元件160‧‧‧Subscription component

125‧‧‧聯絡人列表存庫125‧‧‧Contact List Library

165‧‧‧登錄元件165‧‧‧ Login component

130‧‧‧PSTN網路130‧‧‧PSTN network

170‧‧‧使用者介面元件170‧‧‧User interface components

175‧‧‧客戶端175‧‧‧Client

180‧‧‧電話180‧‧‧Phone

320‧‧‧「Make Call」回應信號320‧‧‧"Make Call" response signal

185‧‧‧傳統PBX185‧‧‧Traditional PBX

190‧‧‧電話190‧‧‧Phone

325‧‧‧「Originated」信號325‧‧‧"Originated" signal

195‧‧‧傳統PBX195‧‧‧Traditional PBX

330‧‧‧「Delivered」信號330‧‧‧"Delivered" signal

200‧‧‧SIP對話建立程序200‧‧‧SIP dialogue establishment procedure

335‧‧‧「Failed」信號335‧‧‧"Failed" signal

205‧‧‧SIP INVITE訊息205‧‧‧SIP INVITE message

340‧‧‧SIP CANCEL訊息340‧‧‧SIP CANCEL message

210‧‧‧SIP 183回應210‧‧‧SIP 183 response

345‧‧‧「Established」信號345‧‧‧"Established" signal

215‧‧‧「Make Call」服務請求信號215‧‧‧"Make Call" service request signal

400‧‧‧INVITE訊息400‧‧‧INVITE message

405‧‧‧IP標頭405‧‧‧IP header

220‧‧‧「Make Call」回應信號220‧‧‧"Make Call" response signal

410‧‧‧TCP標頭410‧‧‧TCP header

412‧‧‧SIP訊息412‧‧‧SIP message

225‧‧‧「Originated」信號225‧‧‧"Originated" signal

415‧‧‧請求列415‧‧‧Request column

230‧‧‧「Delivered」信號230‧‧‧"Delivered" signal

420‧‧‧「From」標頭420‧‧‧"From" header

235‧‧‧「Delivered」信號235‧‧‧"Delivered" signal

425‧‧‧「To」標頭425‧‧‧"To" header

240‧‧‧SIP 180回應240‧‧‧SIP 180 response

430‧‧‧「Caller Tel URI」標頭430‧‧‧"Caller Tel URI" header

245‧‧‧SIP 200 OK回應245‧‧‧SIP 200 OK response

250‧‧‧「Established」信號250‧‧‧"Established" signal

435‧‧‧「Called Tel URI」標頭435‧‧‧"Called Tel URI" header

255‧‧‧「Established」信號255‧‧‧"Established" signal

260‧‧‧SIP ACK260‧‧‧SIP ACK

440‧‧‧「Supplemental Caller ID」標頭440‧‧‧"Supplemental Caller ID" header

300‧‧‧SIP對話建立程序300‧‧‧SIP dialogue establishment procedure

305‧‧‧SIP INVITE訊息305‧‧‧SIP INVITE message

445‧‧‧「Call Type」標頭445‧‧‧"Call Type" header

310‧‧‧SIP 183回應310‧‧‧SIP 183 response

450‧‧‧「Supported」標頭450‧‧‧"Supported" header

315‧‧‧「Make Call」服務請 求信號315‧‧‧"Make Call" service please Signal

455‧‧‧「Content-Length」標 頭455‧‧‧"Content-Length" head

500‧‧‧INVITE回應500‧‧‧INVITE response

532‧‧‧「Called Tel URI」標頭532‧‧‧"Called Tel URI" header

505‧‧‧IP標頭505‧‧‧IP header

510‧‧‧TCP標頭510‧‧‧TCP header

533‧‧‧「Status」標頭533‧‧‧"Status" header

515‧‧‧SIP訊息515‧‧‧SIP message

535‧‧‧「CSeq」標頭535‧‧‧"CSeq" header

520‧‧‧SIP回應識別碼520‧‧‧SIP Response ID

540‧‧‧「Supported」標頭540‧‧‧"Supported" header

525‧‧‧「To」標頭525‧‧‧"To" header

545‧‧‧「Content-Length」標頭545‧‧‧"Content-Length" header

530‧‧‧「From」標頭530‧‧‧"From" header

531‧‧‧「Caller Tel URI」標 頭531‧‧‧"Caller Tel URI" head

第1圖為一方塊圖,其中說明在一具體實施例中,用以將一電話通話關聯於一SIP對話之系統的各元件。Figure 1 is a block diagram illustrating elements of a system for associating a telephone call to a SIP dialog in a particular embodiment.

第2圖為一網路作業(networking)圖,其中顯示 一在一具體實施例中,用以關聯於一電話通話而建立起一SIP對話的通訊流程。Figure 2 shows a networking diagram showing In a specific embodiment, a communication flow for establishing a SIP conversation is associated with a telephone call.

第3圖為一網路作業圖,其中顯示在一存有競賽情況的條件下,用以關聯於一電話通話而建立起一SIP對話的通訊流程。Figure 3 is a network operation diagram showing the communication flow for establishing a SIP conversation associated with a telephone call in the presence of a competition situation.

第4圖顯示在一具體實施例中,一自一發話方客戶端所發送之SIP INVITE訊息的內容。Figure 4 shows the contents of a SIP INVITE message sent from a client client in a particular embodiment.

第5圖顯示在一具體實施例中,一自一受話方客戶端所發送之SIP INVITE回應訊息的內容。Figure 5 shows the contents of a SIP INVITE response message sent from a recipient client in a particular embodiment.

第6圖為一流程圖,其中說明在一具體實施例中之例式,其係在一發話者計算裝置處建立起一關聯於一電話通話之SIP對話。Figure 6 is a flow diagram illustrating an example in a particular embodiment for establishing a SIP dialog associated with a telephone call at a talker computing device.

第7圖為一流程圖,其中說明在一具體實施例中之例式,其係在一受話方計算裝置處建立起一關聯於一電話通話之SIP對話的例式。Figure 7 is a flow diagram illustrating an example in a particular embodiment for establishing a SIP dialog associated with a telephone call at a subscriber computing device.

100‧‧‧作用中通訊伺服器100‧‧‧Active communication server

105‧‧‧存現伺服器105‧‧‧Current server

110‧‧‧使用者資料伺服器110‧‧‧User data server

115‧‧‧訂閱元件115‧‧‧Subscription component

120‧‧‧存現文件存庫120‧‧‧Current file repository

125‧‧‧聯絡人列表存庫125‧‧‧Contact List Library

130‧‧‧PSTN網路130‧‧‧PSTN network

135‧‧‧網際網路135‧‧‧ Internet

150‧‧‧客戶端A150‧‧‧Client A

175‧‧‧客戶端B175‧‧‧Client B

155‧‧‧通訊應用程式155‧‧‧Communication application

160‧‧‧訂閱元件160‧‧‧Subscription component

165‧‧‧登錄元件165‧‧‧ Login component

170‧‧‧使用者介面元件170‧‧‧User interface components

175‧‧‧客戶端175‧‧‧Client

180‧‧‧電話180‧‧‧Phone

185‧‧‧傳統PBX185‧‧‧Traditional PBX

190‧‧‧電話190‧‧‧Phone

195‧‧‧傳統PBX195‧‧‧Traditional PBX

Claims (20)

一種儲存電腦可執行指令之電腦可讀取儲存記憶體,該等電腦可執行指令用於控制一受話者(callee)之一裝置,以提供建構在一公眾電話網路上之電話通話通知,該等電腦可執行指令用於控制該受話者之該裝置以實行一方法,該方法包含以下步驟:經由相對於該公眾電話網路為帶外(out-of-band)之一電腦網路接收一第一訊息,該第一訊息表示一發話者(caller)想要經由該公眾電話網路啟動與該受話者之一電話通話;在接收該第一訊息之後,經由該電腦網路發送一回應,該回應表示該受話者之該裝置已接收該第一訊息;以及在經由該公眾電話網路傳遞該電話通話至該受話者之後,經由該電腦網路接收一第二訊息,該第二訊息表示已傳遞至該受話者之該電話通話係來自該發話者;以及在接收該第二訊息之後,通知該受話者該傳遞之電話通話係來自該發話者。 A computer readable storage memory for storing computer executable instructions for controlling a device of a callee to provide a telephone call notification constructed on a public telephone network, such Computer-executable instructions for controlling the device of the callee to perform a method, the method comprising the steps of: receiving a first computer network out-of-band with respect to the public telephone network a message indicating that a caller wants to initiate a telephone conversation with one of the recipients via the public telephone network; after receiving the first message, sending a response via the computer network, Responding to indicate that the device of the callee has received the first message; and after transmitting the call to the callee via the public telephone network, receiving a second message via the computer network, the second message indicating that The telephone conversation delivered to the callee is from the talker; and after receiving the second message, the caller is notified that the delivered telephone call is from the talker. 如請求項第1項所述之電腦可讀取儲存記憶體,其中該第一訊息與該第二訊息係從該發話者之 一裝置發送。 The computer readable storage memory of claim 1, wherein the first message and the second message are from the Talker A device sends. 如請求項第2項所述之電腦可讀取儲存記憶體,其中在該發話者之該裝置接收該電話通話已傳遞至該受話者之通知之後,發送該第二訊息。 The computer-readable storage memory of claim 2, wherein the second message is sent after the device of the caller receives the notification that the telephone call has been delivered to the callee. 如請求項第1項所述之電腦可讀取儲存記憶體,其中在被通知該電話通話係來自該發話者之後,該受話者應答該電話通話。 The computer readable storage memory of claim 1, wherein the callee answers the telephone call after being notified that the telephone conversation is from the Talker. 如請求項第1項所述之電腦可讀取儲存記憶體,其中在被通知該電話通話係來自該發話者之前,該受話者應答該電話通話。 The computer readable storage memory of claim 1 wherein the recipient answers the telephone conversation before being notified that the telephone conversation is from the Talker. 如請求項第1項所述之電腦可讀取儲存記憶體,其中該第一訊息、該回應與該第二訊息符合一對話啟動協定。 The computer-readable storage memory of claim 1, wherein the first message, the response, and the second message conform to a dialog initiation protocol. 如請求項第1項所述之電腦可讀取儲存記憶體,其中該方法進一步包含以下步驟:在接收該第一訊息之後,通知該受話者以預期將傳遞來自該發話者之一電話通話。 The computer readable storage memory of claim 1, wherein the method further comprises the step of, after receiving the first message, notifying the recipient that a telephone call from one of the talkers is expected to be delivered. 如請求項第1項所述之電腦可讀取儲存記憶體,其中該方法進一步包含以下步驟:在傳遞該電話通話之後,經由該電腦網路發送一第二回應,該第二回應表示一電話通話已傳遞至該受話者。 The computer readable storage memory according to claim 1, wherein the method further comprises the step of: after transmitting the telephone call, sending a second response via the computer network, the second response indicating a call The call has been delivered to the callee. 一種儲存電腦可執行指令之電腦可讀取儲存記憶體,該等電腦可執行指令用於控制一受話者之一裝置,以提供建構在一公眾電話網路上之從一發話者至該受話者之一電話通話通知,該等電腦可執行指令用於控制該受話者之該裝置以實行一方法,該方法包含以下步驟:經由相對於該公眾電話網路為帶外之一電腦網路接收一訊息,而在該發話者之一裝置接收該電話通話已經由該公眾電話網路傳遞至該受話者之後,藉由該發話者之該裝置發送該訊息,且該訊息表示該電話通話係來自該發話者;以及在接收該訊息之後,通知該受話者該傳遞之電話通話係來自該發話者,以使得該受話者在應答該電話通話之前知道該發話者之該識別。 A computer readable storage memory for storing computer executable instructions for controlling a device of a callee to provide a caller from the caller to the caller on a public telephone network a telephone call notification, the computer executable instructions for controlling the device of the callee to perform a method, the method comprising the steps of: receiving a message for a network of out-of-band computers relative to the public telephone network And after the device of the Talker receives the call that has been transmitted by the public telephone network to the callee, the device sends the message by the Talker, and the message indicates that the call is from the utterance And after receiving the message, the caller is notified that the delivered call is from the caller so that the callee knows the identity of the caller before answering the call. 如請求項第9項所述之電腦可讀取儲存記憶體,其中該方法進一步包含以下步驟: 經由該電腦網路接收一第一訊息,該第一訊息表示該發話者想要經由該公眾電話網路啟動與該受話者之該電話通話;在接收該第一訊息之後,經由該電腦網路發送一回應,該回應表示該受話者之該裝置已接收該第一訊息。 The computer readable storage memory according to claim 9, wherein the method further comprises the following steps: Receiving a first message via the computer network, the first message indicating that the caller wants to initiate a call with the callee via the public telephone network; after receiving the first message, via the computer network Sending a response indicating that the device of the callee has received the first message. 如請求項第9項所述之電腦可讀取儲存記憶體,其中該訊息與該回應符合一對話啟動協定。 The computer readable storage memory as described in claim 9 wherein the message conforms to the response initiation agreement. 如請求項第9項所述之電腦可讀取儲存記憶體,其中該方法進一步包含以下步驟:在傳遞該電話通話之後,經由該電腦網路發送一回應,該回應表示一電話通話已傳遞至該受話者。 The computer readable storage memory according to claim 9, wherein the method further comprises the step of: after transmitting the telephone call, sending a response via the computer network, the response indicating that a telephone call has been delivered to The recipient. 一種用於提供從一發話者至一受話者之一電話通話之通知的該受話者之裝置,該電話通話建構在一公眾切換電話網路上,該受話者之該裝置包含:一儲存記憶體,該儲存記憶體儲存電腦可執行指令,該等電腦可執行指令包括:一接收預傳遞訊息部件,該接收預傳遞訊息部件經由相對於該公眾切換電話網路為帶外之一電腦網路一電腦網路接收一預傳遞訊息,該預傳遞訊息 表示該發話者想要經由該公眾切換電話網路啟動與該受話者之一電話通話;一發送回應部件,該發送回應部件在接收該預傳遞訊息之後,經由該電腦網路發送一回應,該回應表示該受話者之該裝置已接收該後傳遞訊息;以及一接收後傳遞訊息部件,該接收後傳遞訊息部件經由該電腦網路接收一後傳遞訊息,該後傳遞訊息表示來自該發話者之該電話通話已傳遞至該受話者;以及一第一通知部件,該第一通知部件在接收該後傳遞訊息之後,通知該受話者該傳遞之電話通話係來自該發話者;以及一處理器,該處理器用於執行儲存在該儲存記憶體上的該等電腦可執行指令。 A device for providing a call from a caller to a caller of a caller, the call being constructed on a public switched telephone network, the device comprising: a storage memory, The storage memory stores computer executable instructions, the computer executable instructions comprising: a receiving pre-delivery message component, the receiving pre-delivery message component being a computer network outside the band via the public switched telephone network The network receives a pre-delivery message, the pre-delivery message Representing that the caller wants to initiate a telephone conversation with one of the recipients via the public switched telephone network; a sending response component that sends a response via the computer network after receiving the pre-delivery message, Responding to indicate that the device of the callee has received the post-delivery message; and a post-received message component, the post-received message component receives a post-delivery message via the computer network, and the post-delivery message indicates from the caller The telephone call has been delivered to the callee; and a first notification component, after receiving the post-delivery message, notifying the callee that the delivered telephone call is from the caller; and a processor, The processor is operative to execute the computer executable instructions stored on the storage memory. 如請求項第13項所述之裝置,其中該預傳遞訊息與該後傳遞訊息係從該發話者之裝置發送。 The device of claim 13, wherein the pre-delivery message and the post-delivery message are transmitted from the Talker device. 如請求項第13項所述之裝置,其中在該發話者之該裝置接收該電話通話已傳遞至該受話者之通知之後,發送該後傳遞訊息。 The device of claim 13, wherein the post-delivery message is sent after the device of the caller receives the notification that the telephone call has been delivered to the callee. 如請求項第13項所述之裝置,其中在被通知該電話通話係來自該發話者之後,該受話者應答該電話通話。 The device of claim 13, wherein the callee answers the telephone call after being notified that the telephone conversation is from the talker. 如請求項第13項所述之裝置,其中在被通知該電話通話係來自該發話者之前,該受話者應答該電話通話。 The device of claim 13, wherein the callee answers the telephone call before being notified that the telephone conversation is from the talker. 如請求項第13項所述之裝置,其中該預傳遞訊息、該回應與該後傳遞訊息符合一對話啟動協定。 The device of claim 13, wherein the pre-delivery message, the response, and the post-delivery message conform to a dialog initiation protocol. 如請求項第13項所述之裝置,進一步包括:一第二通知部件,該第二通知部件在接收該預傳遞訊息之後,通知該受話者以預期將傳遞來自該發話者之一電話通話。 The device of claim 13, further comprising: a second notification component, after receiving the pre-delivery message, notifying the recipient that an incoming call from the caller is expected to be delivered. 如請求項第13項所述之裝置,進一步包括:一發送第二回應部件,該發送第二回應部件在傳遞該電話通話之後,經由該電腦網路發送一第二回應,該第二回應表示一電話通話已傳遞至該受話者。 The device of claim 13, further comprising: transmitting a second response component, the transmitting second response component transmitting a second response via the computer network after transmitting the telephone call, the second response indicating A phone call has been delivered to the callee.
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